asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	1a00cbbf1d	Fixes issue with AST_CONTROL_SRCUPDATE breaking out of native bridge In iax2, when a AST_CONTROL_SRCUPDATE is received it breaks from the native bridge, but since there is no code path to handle srcupdate it just goes to be beginning of the loop. This was causing packet storms of srcupdate frames between servers. Now srcupdate frames do not break the native bridge for processing. (closes issue #13749) Reported by: adiemus git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	1b5b3efcae	fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	22734e39dc	Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	db4dc67740	fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	4c3b9ccf3b	format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	beaf6760c4	Disable format_ilbc.so by default, like codec_ilbc.so git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	ada21a8039	Make sure that the debug line is not printed on debug level 0 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	3cc3863d28	Zaptel is not DAHDI. Rather, Zaptel is actually Zaptel. (in case you're confused, DAHDI is still DAHDI) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	3e0b36d9d0	Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7accd1ec46	Fix a place where filestreams were not refcounted properly This section was already present in trunk and other branches, but did not exist in 1.4. (closes issue #14395) Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut (license 60) Tested by: ZX81 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	a13deff994	Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	46963bc8b5	Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	3e396d458a	Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	bfaa341f58	Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Philippe Sultan	d045d36561	Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	70f7c7e9cb	Revert RTP changes for continuation of DTMF. Proxy commit by russell via SMS. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1d4e4ff3d1	Clear out the current event after forcing the end of a digit git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	9dff8995b4	Fixify infinite DTMF in the case that no RFC2833 END event is ever received git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	9e38dd5427	Restore a behavior that was recently changed, when we fixed issue #13962 and issue #13363 (related to issue #6176). When a hangup occurs during a Macro execution in earlier 1.4, the h extension would execute within the Macro context, whereas it was always supposed to execute only within the main context (where Macro was called). So this fix checks for an "h" extension in the deepest macro context where a hangup occurred; if it exists, that "h" extension executes, otherwise the main context "h" is executed. (closes issue #14122) Reported by: wetwired Patches: 20090210__bug14122.diff.txt uploaded by Corydon76 (license 14) Tested by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	9fa3324845	Go off hold when we get an empty reinvite telling us to. (closes issue #14448) Reported by: frawd Patches: hold_invite_nosdp.patch uploaded by frawd (license 610) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Nicholson	b60ec2baa1	Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	680cc35607	This patch solves some compiler complaints in both 32 and 64-bit environments. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7f20e5ffab	Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e2fd8852db	Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c347a43c9f	Fix a race condition that could cause a crash. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Dwayne M. Hubbard	d29a99cb89	check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	c80b2b93b5	Remove a debug message I put in by accident. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	6cda579f17	Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Nicholson	5edf9d8a59	Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	76af76c5f4	Backport OS X fix from trunk (AGAIN, closes issue #14360) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	f70845aa24	Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests With this fix, we only will perform an SRV lookup at the following times: * The first time we register with a remote registrar * If we send a REGISTER but do not receive a response * If the sendto() function returns an error While I wrote the patch that fixes this issue, a huge amount of credit is due to Brett Bryant, who wrote the initial patch on which I based this one. (closes issue #12312) Reported by: jrast Patches: 12312.patch uploaded by putnopvut (license 60) Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	0b026ce04f	Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	e7478c7b15	Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	d914df762a	Add some missing cleanup to app_mixmonitor git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8e764c583d	Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	82937be553	Revert my previous change because it was stupid git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	fed7d2308b	Add a missing unlock. Extremely unlikely to ever matter, but it's needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	28056ffc94	Fixes issue with IAX2 transfer not handing off calls. Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	87c921a68e	Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	13138151e1	Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	1a9018e799	Fix a feature inheritance bug I added after code review git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Richard Mudgett	cefe4f025d	channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	2b340eab54	Rename new parkedcallparking option to parkedcallreparking Since this option actually already existed in 1.6.0+, use the same name so as not to confuse people when they upgrade git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	4e069885ce	Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c257ffeed0	Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	3209942b7e	Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	13a60eba0c	This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	16f378c559	Clarify log message (suggested by manxpower on #asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	566429c300	Add a better explanation of the difference between the device namespace and the dialplan for newbies. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	cade7e1559	Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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